Telephone exchanges of the prior art may be located in house at a subscriber's office, or in a remote terminal, or in a central office of a telephony service provider. Such a telephone exchange normally interconnects one telephone to another telephone if directly connected to both telephones, and connects any directly connected telephone to an outside telephone network (also called wide area network abbreviated as “WAN”). Conventional telephone exchanges normally support analog phones (that require circuit-switching), although recently a telephone may also be used to place and receive voice over Internet Protocol (VoIP)-based telephone calls and public switched telephone network (PSTN)-based telephone calls, as described in, for example, U.S. Pat. No. 6,141,341 granted to Jones, et al. that is incorporated by reference herein in its entirety. Internet Protocol (IP) based telephony is also described in, for example, U.S. Pat. No. 6,252,952 granted to Kung, et al. and U.S. Pat. No. 5,742,596 granted to Baratz, et al. each of which is incorporated by reference herein in its entirety.
Telephone exchanges that are located in a subscriber's office (called private branch exchanges, abbreviated as “PBX”) typically perform various telephone management functions, such as call transfer, call forwarding, speed dialing, music on hold, last number re-dial, 3-way conferencing, call blocking, call accounting, as well as integrating voice mail systems, and facsimile machines, although such functions may also be provided by telephone exchanges located in the central office (also called “central office switch”). An example of a packet-switching PBX is described in, for example, “Intranet and IP-Based UnPBXs,” Chapter 7, pages 7-16 to 7-22 in the book entitled “the UnPBX” edited by Edwin Margulies, Flatiron Publishing, Inc. 1997 that is incorporated by reference herein in its entirety.
See also, the article entitled “Design and Evaluation of an Extensible Web & Telephony Server based on the J-Kernel” by Daniel Spoonhower et al., Technical Report 98-1715, Department of Computer Science, Cornell University, November 1998 available on the Internet at http://www.cs.cornell.edu/Info/People/chichao/tr1715.pdf that is incorporated by reference herein in its entirety. Another article entitled “Towards Junking the PBX: Deploying IP Telephony” by Wenyu Jiang, et al. NOSSDAV '01, Jun. 25-27, 2001, Port Jefferson, N.Y., is also incorporated by reference herein in its entirety. Telephone exchanges for IP phones may conform to any of the following protocols: H.323, SIP and MGCP. These protocols are well known in the art and are described at the following webpages:
For H.323 protocol:
http://www.itu.int/rec/recommendation.asp?type=folders&lang=e
&parent=T-REC-H.323
For SIP protocol:
http://www.ietf.org/rfc.html
see RFC 2543bis
For MGCP protocol:
http://www.ietf.org/rfc.html
see RFC 2705bis
Traditionally, before a phone is used with a PBX or with a central office switch, one or more parameters that govern telephony-related services to be provided to a user of the phone must be specified by a human (hereinafter “administrator”). Specifically, before a user receives any telephony-related service (hereinafter simply “service”) at a phone, the administrator associates a telephone number (unique within the telephone exchange) with a port of the telephone exchange to which the phone is coupled, only then does the telephone exchange route calls to the phone (using this association). An administrator manually performs this association (in a process called provisioning) at a computer for managing the telephone exchange (which may be called, for example, “Operations Support System” abbreviated as OSS). Such a computer may also be known as, for example, Headquarters Server.
During provisioning, the administrator typically also performs other tasks, such as enabling voice mail service for calls to the port that are not answered, enabling access from the port to the WAN, and enabling access from the port to various services such as speed calling, call waiting, and call forwarding services. Information provided by the administrator is typically stored in a record of a database which contains a number of such records, one per port. After creation of such a record, the full range of services that are provisioned (including dial tone) are activated (in a process called “service activation”), e.g. by appropriately configuring the telephone exchange. Thereafter, as soon as a phone is coupled to the newly-provisioned port, the phone is provided with access to the enabled services. In such systems, a user cannot receive any service until after their port has been provisioned, and service activated.
U.S. Pat. No. 5,588,051 granted to Berkowitz et al (which is incorporated by reference herein in its entirety) describes associating unsubscribed lines with the same telephone number and same switch line termination point. The unsubscribed lines may be inexpensively provisioned with soft dial tone thereby allowing a potential telephone subscriber to interactively subscribe to and activate telephone service from an unsubscribed line. Soft dial tone is also known in the art as limited access dial tone, left-in dial tone, quick serve, or warm line service. Since soft dial tone is not a billed service, calls directed to the unsubscribed telephone line are blocked at the central office switch. The application of soft dial tone may be used to identify a network access endpoint that is being used by a customer, and thus improve the accuracy of the contents of databases used by the operations facility in the provisioning of telephone service by identifying incorrect data.
U.S. Pat. No. 5,953,389 granted to Pruett et al (which is incorporated by reference herein in its entirety) describes receiving a service request from a customer at a living unit requesting service, and determining if the service request is eligible for processing by an adjunct processor. When eligible, the method automatically provisions customer equipment to execute the service request based on customer identification data and customer facilities.
A reference called “Cisco CallManager System Guide” describes, in Chapter 11, a method called “Auto-Registration” that automatically assigns directory numbers to new devices as they connect to the IP telephony network. Auto-registration is to be used if the administrator wants Cisco CallManager to assign directory numbers automatically to new phones when the phones are plugged into the network. Since a directory number is assigned to a new device, it appears (to the inventors) that phone calls can be received at the new device.
Cisco's CallManager disables auto-registration by default. Enabling auto-registration carries a security risk in that “rogue” phones can automatically register with Cisco CallManager. Accordingly, it is suggested by Cisco that auto-registration should be enabled only for brief periods when the administrator wants to perform bulk phone adds. After a new device has been auto-registered, Cisco requires an administrator to assign a user to the new device, via a graphical user interface.